This is correct, if no NAT is involved anywhere and reinvites are allowed then Asterisk will stay out of the media path and be used only as Signaling server. So as for your answer yes, it will be able to handle more calls than expected because there is no CPU overhead of the media path.

It is common strategy to have a single signaling server and have RTP servers all around the globe for latency and etc, media gateways.

Vicky wrote:
Asterisk wont sit in media path if both callee and caller agrees on common codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan ( please correct me if i am wrong ) , no call recording is enabled . I think asterisk does native bridging even if one is behind nat ( i tested with atleast one party behind nat not sure if it works when both are behind nat ) and devices should support reinvites ..

On 03/12/06, *Dovid B* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:

    I know this has been asked before and I went over the wiki but I
    have not been able to come to a clear answer.
1) If I have SIP Provider ----> Asterisk -----> ATA and vice versa
    (ATA -----> Asterisk ----> SIP Provider) from what I understand if
    NO NAT is being used then asterisk just starts and stops the
    session however the RTP media stream will be passed directly from
    the SIP provider and vice versa. (This is of course if there is no
    NAT involved). Now say I had such a set up will the server be able
    to handle more calls than "average" if the only responsibility if
    the server is to authenticated and pass along the calls ? (There
    will be an AGI running in the begining to determine what route to
    used based on how many minutes each route has used). Now if the
    ATA's are behind VOIP and asterisk is on a public IP then does
    asterisk have to sit in the media path ? Also can some one explain
    exaclty when the RTP session is started and stopped.
Also another set up we are woroking on is SIP Provider (Incoming
    DID)  ----> Asterisk (for authentication based on PIN) -----> Back
    to SIP Provider. The asterisk server will be on a public IP. Can I
    have asterisk stay out fo the media path (here I asume yes. Just
    wana be 100% sure).
Thanks a lot. Dovid

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